Cisco Unified Communications Manager Express (CUCME or CME) is a software-based IP PBX that runs on Cisco routers. It provides basic call processing, routing, and IP telephony enablement functions. CME supports both SCCP and SIP protocols. Key features of CME include local directory service, call forwarding, call transfer, call park, pickup groups, phone templates, paging, single number reach for mobility, and hunt groups. Phones connect to CME via SCCP or SIP and are configured with directory numbers, buttons, and other settings. CME provides a cost-effective IP PBX solution for small-to-medium sized businesses.
CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course
CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course
CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course
CCIE Collaboration Lecture Chapter 9.3 cucm mobility single number reach an...Faisal Khan
CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course
SPARSH VP248 is a high-definition VoIP phone built with superior acoustics and elegant design to provide unsurpassed audio quality and rich user experience.
Based on open-standard SIP protocol, SPARSH VP248 is interoperable with any standard SIP infrastructure such as IP-PBX, SIP Proxies, Softswitches and Stand-alone applications.
SPARSH VP248 is designed for power users, knowledge workers and managers for quick access totheadvance system features and functions. A feature-packed IP phoneenables user to work efficiently with advance call handling capabilities
CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course
CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course
CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course
CCIE Collaboration Lecture Chapter 9.3 cucm mobility single number reach an...Faisal Khan
CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course
SPARSH VP248 is a high-definition VoIP phone built with superior acoustics and elegant design to provide unsurpassed audio quality and rich user experience.
Based on open-standard SIP protocol, SPARSH VP248 is interoperable with any standard SIP infrastructure such as IP-PBX, SIP Proxies, Softswitches and Stand-alone applications.
SPARSH VP248 is designed for power users, knowledge workers and managers for quick access totheadvance system features and functions. A feature-packed IP phoneenables user to work efficiently with advance call handling capabilities
Matrix feature-rich ATAs offer connectivity to VoIP, GSM and POTS networks. An ATA user can plug standard analog telephone devices to the ATA and the analog device(s) will connect transparently to the IP and GSM networks. An ATA thus provides a user with the ease of using a standard telephone instrument, yet make VoIP and GSM calls. The ATAs can also be interfaced to existing PBX system, offering GSM and IP line to be shared among the PBX users.
Matrix Telecom Solutions: SETU VTEP - Fixed VoIP to T1/E1 PRI GatewayMatrix Comsec
Matrix SETU VTEP is a compact and dedicated gateway for VoIP to T1/E1 PRI network offering high-value communication experience to businesses of all size, Service Providers, Call Centers and simple but cost-effective solution for multi-location branch office communication. This intelligently designed gateway incorporates advanced features with multiple connectivity options to connect with a legacy communication system using T1/E1 or PRI signaling. SETU VTEP offers reliable and cost-effective solutions to the changing requirements of the business communication and offer customer value for money.
Matrix Telecom Solutions: ETERNITY NE - IP-PBXMatrix Comsec
Matrix Presents, ETERNITY NE – The Next Generation IP-PBX built to mature today’s nascent businesses into flourished enterprises, breaking the confines of conventional telephony systems. Right from its perception, each and every attribute of ETERNITY NE revolves around the needs of the small businesses. ETERNITY NE brings complete synergy of the wired, wireless and packet based communication into the SMB premise. The multi-functional persona of NE wipes out the need of investing in multiple devices. With the superior features that it delivers, it puts any small business at par to compete actively with the constantly evolving and challenging surrounding.
Matrix Telecom Solutions: NAVAN CNX200 - Office-in-a-Box Solution for Small B...Matrix Comsec
NAVAN CNX200 is an all-in-one office solution for small businesses and enterprise branch offices with up to 24 users. It combines the functionalities of IP-PBX, Data router, Wi-Fi access point, VoIP-GSM gateway, VPN and Firewall Security in a compact and converged platform. A true office-in-a-box, CNX200 innovates the way small businesses communicate and manage infrastructure, so that they can increase productivity, lower costs and enhance collaboration with customers and suppliers.
Telephone System, Phone System, Business Phone System, VoIP Phone System, Hos...Rob Bliss
Zultys has been recognized as one of the leaders in the IP Telephone System Industry. Selected as the "2009 Internet Telephony" for it's Recognition as a leader in IP Telephone Systems. This is not handed out to any manufacturer, but the ones who like Zultys have continuiously shown consecutive improvement year after year.
Matrix Telecom Solutions: SETU VGFX - Fixed VoIP to GSM/3G-FXO-FXS Voice Gat...Matrix Comsec
Matrix presents SETU VGFX- The Single-box Gateway solution, offering seamless connectivity between VoIP, GSM and POTS (FXO and FXS) networks. SETU VGFX supports flexible and intelligent call routing options to ensure that communication always happens through the most cost effective network.
Matrix Telecom Solutions: SETU VFXTH - Fixed VoIP to FXO-FXS GatewaysMatrix Comsec
Matrix presents SETU VFXTH-The multi-channel SIP gateway offering seamless connectivity between VoIP and PSTN networks through multiple FXS and FXO ports. Matrix SETU VFXTH offers universal and transparent call routing irrespective of type of ports – VoIP-FXS, VoIP-FXO and FXS-FXO. Its superior call and signal processing capabilities ensure unrestricted flow of multiple calls with higher speed and better speech quality.
This is the Standard User Demonstration of Melbourne University's IP Telephony Service. We recommend all staff moving to the University's IP Telephony system review this presentation
Matrix feature-rich ATAs offer connectivity to VoIP, GSM and POTS networks. An ATA user can plug standard analog telephone devices to the ATA and the analog device(s) will connect transparently to the IP and GSM networks. An ATA thus provides a user with the ease of using a standard telephone instrument, yet make VoIP and GSM calls. The ATAs can also be interfaced to existing PBX system, offering GSM and IP line to be shared among the PBX users.
Matrix Telecom Solutions: SETU VTEP - Fixed VoIP to T1/E1 PRI GatewayMatrix Comsec
Matrix SETU VTEP is a compact and dedicated gateway for VoIP to T1/E1 PRI network offering high-value communication experience to businesses of all size, Service Providers, Call Centers and simple but cost-effective solution for multi-location branch office communication. This intelligently designed gateway incorporates advanced features with multiple connectivity options to connect with a legacy communication system using T1/E1 or PRI signaling. SETU VTEP offers reliable and cost-effective solutions to the changing requirements of the business communication and offer customer value for money.
Matrix Telecom Solutions: ETERNITY NE - IP-PBXMatrix Comsec
Matrix Presents, ETERNITY NE – The Next Generation IP-PBX built to mature today’s nascent businesses into flourished enterprises, breaking the confines of conventional telephony systems. Right from its perception, each and every attribute of ETERNITY NE revolves around the needs of the small businesses. ETERNITY NE brings complete synergy of the wired, wireless and packet based communication into the SMB premise. The multi-functional persona of NE wipes out the need of investing in multiple devices. With the superior features that it delivers, it puts any small business at par to compete actively with the constantly evolving and challenging surrounding.
Matrix Telecom Solutions: NAVAN CNX200 - Office-in-a-Box Solution for Small B...Matrix Comsec
NAVAN CNX200 is an all-in-one office solution for small businesses and enterprise branch offices with up to 24 users. It combines the functionalities of IP-PBX, Data router, Wi-Fi access point, VoIP-GSM gateway, VPN and Firewall Security in a compact and converged platform. A true office-in-a-box, CNX200 innovates the way small businesses communicate and manage infrastructure, so that they can increase productivity, lower costs and enhance collaboration with customers and suppliers.
Telephone System, Phone System, Business Phone System, VoIP Phone System, Hos...Rob Bliss
Zultys has been recognized as one of the leaders in the IP Telephone System Industry. Selected as the "2009 Internet Telephony" for it's Recognition as a leader in IP Telephone Systems. This is not handed out to any manufacturer, but the ones who like Zultys have continuiously shown consecutive improvement year after year.
Matrix Telecom Solutions: SETU VGFX - Fixed VoIP to GSM/3G-FXO-FXS Voice Gat...Matrix Comsec
Matrix presents SETU VGFX- The Single-box Gateway solution, offering seamless connectivity between VoIP, GSM and POTS (FXO and FXS) networks. SETU VGFX supports flexible and intelligent call routing options to ensure that communication always happens through the most cost effective network.
Matrix Telecom Solutions: SETU VFXTH - Fixed VoIP to FXO-FXS GatewaysMatrix Comsec
Matrix presents SETU VFXTH-The multi-channel SIP gateway offering seamless connectivity between VoIP and PSTN networks through multiple FXS and FXO ports. Matrix SETU VFXTH offers universal and transparent call routing irrespective of type of ports – VoIP-FXS, VoIP-FXO and FXS-FXO. Its superior call and signal processing capabilities ensure unrestricted flow of multiple calls with higher speed and better speech quality.
This is the Standard User Demonstration of Melbourne University's IP Telephony Service. We recommend all staff moving to the University's IP Telephony system review this presentation
Matrix ETERNITY is a family of IP-PBXs with Universal Connectivity and Seamless Mobility. The ETERNITY IP-PBX offers built-in gateway capability to connect nearly all telecom interfaces like FXS, FXO, ISDN BRI, ISDN PRI, T1/E1, GSM and 3G.
TelePacific Communications Hosted Telephone System (PBX) presentation shows why this offering is different from the rest. With over 40,000 business customers, this 600M CLEC supports it's business customers with 97% satisfaction rating.
Benefit from IP communications, easier network management, and enhanced employee productivity with our end-to-end IP communication solution from the leader in business Voice over IP and SIP trunking services.
Voice over Internet Protocol (Voice over IP, VoIP and IP telephony) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).The steps and principals involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding.
Microsoft Teams
Built for hybrid work
Feel seen and heard no matter where you are and do your best work, your way.
Empower people and teams in a hybrid work world
Flexible work is here to stay, and embracing it is critical to your future success. The latest tools from Microsoft empower hybrid work environments and enable flexible work beyond organizational boundaries.
Fully featured Polycom Desktop and Conference IP telephones that work in conjunction with the NBS Voice over Internet Communications Exchange (NBS VoICE) cloud-based hosted voice service.
Similar to Ccnp collaboration plus module 1 chapter 5 cisco unified communication express (20)
Biological screening of herbal drugs: Introduction and Need for
Phyto-Pharmacological Screening, New Strategies for evaluating
Natural Products, In vitro evaluation techniques for Antioxidants, Antimicrobial and Anticancer drugs. In vivo evaluation techniques
for Anti-inflammatory, Antiulcer, Anticancer, Wound healing, Antidiabetic, Hepatoprotective, Cardio protective, Diuretics and
Antifertility, Toxicity studies as per OECD guidelines
Macroeconomics- Movie Location
This will be used as part of your Personal Professional Portfolio once graded.
Objective:
Prepare a presentation or a paper using research, basic comparative analysis, data organization and application of economic information. You will make an informed assessment of an economic climate outside of the United States to accomplish an entertainment industry objective.
Introduction to AI for Nonprofits with Tapp NetworkTechSoup
Dive into the world of AI! Experts Jon Hill and Tareq Monaur will guide you through AI's role in enhancing nonprofit websites and basic marketing strategies, making it easy to understand and apply.
2024.06.01 Introducing a competency framework for languag learning materials ...Sandy Millin
http://sandymillin.wordpress.com/iateflwebinar2024
Published classroom materials form the basis of syllabuses, drive teacher professional development, and have a potentially huge influence on learners, teachers and education systems. All teachers also create their own materials, whether a few sentences on a blackboard, a highly-structured fully-realised online course, or anything in between. Despite this, the knowledge and skills needed to create effective language learning materials are rarely part of teacher training, and are mostly learnt by trial and error.
Knowledge and skills frameworks, generally called competency frameworks, for ELT teachers, trainers and managers have existed for a few years now. However, until I created one for my MA dissertation, there wasn’t one drawing together what we need to know and do to be able to effectively produce language learning materials.
This webinar will introduce you to my framework, highlighting the key competencies I identified from my research. It will also show how anybody involved in language teaching (any language, not just English!), teacher training, managing schools or developing language learning materials can benefit from using the framework.
Model Attribute Check Company Auto PropertyCeline George
In Odoo, the multi-company feature allows you to manage multiple companies within a single Odoo database instance. Each company can have its own configurations while still sharing common resources such as products, customers, and suppliers.
Operation “Blue Star” is the only event in the history of Independent India where the state went into war with its own people. Even after about 40 years it is not clear if it was culmination of states anger over people of the region, a political game of power or start of dictatorial chapter in the democratic setup.
The people of Punjab felt alienated from main stream due to denial of their just demands during a long democratic struggle since independence. As it happen all over the word, it led to militant struggle with great loss of lives of military, police and civilian personnel. Killing of Indira Gandhi and massacre of innocent Sikhs in Delhi and other India cities was also associated with this movement.
3. 333
Overview
• IOS feature that turns Cisco Router a IP Based PBX
• Call processing and device control
• Command-line or GUI-based configuration
• Local directory service
• Computer Telephony Integration (CTI) support
• Trunking to other VoIP systems
• Direct integration with Cisco Unity Express (CUE)
PSTN
IP WAN
4. 444
Setting up CME for SCCP & SIP
Based on SCCP
telephony-service
max-ephones 15
max-dn 15
ip source-address 142.102.65.254 port 2000
max-conferences 4
dn-webedit
time-webedit
transfer-system full-consult
voice service voip
allow-connection sip to sip
sip
bind control source-interface GigabitEthernet0/0.102
bind media source-interface GigabitEthernet0/0.102
registrar server expires max 1200 min 300
voice register global
mode cme
source-address 142.102.66.254 port 5060
max-dn 10
max-pool 10
authenticate register
authenticate realm voicebootcamp.com
create profile sync 0041377162820495
ntp-server 135.11.11.11 mode unicast
Based on SIP Protocol
Based on SCCP
5. 555
Phones in Cisco Unified CME
Directory Number – EPHONE-DN
• A directory number, also known as an
ephone-dn for SCCP that represents the line
connecting a voice channel to a phone
• Each directory number has a unique dn-tag
• Directory numbers are assigned to line
buttons on phones during configuration
• One virtual voice port and one or more dial
peers are automatically created for each
directory number
• Number of simultaneous calls that you can
have
• Not all types of directory numbers can be
configured for all phones or for all
protocols
ephone-dn 2
number 6001
ephone 1
Mac-
button 1:2
ephone-dn 3 dual-line
number 6002
ephone-dn 4 octo-line
number 6003
ephone 2
Mac-address Y.Y.Y.y
button 1:3 2:4
Single Line
6. 666
Phones in Cisco Unified CME
Directory Number – EPHONE-DN
• Following Type of Directory Number Exist
• Single-Line
• Dual-Line
• Octo-Line
• Shared Line (Exclusive)
• Mixed Shared Lines
• Overlaid
ephone-dn 2
number 6001
ephone 1
Mac-
button 1:2
ephone-dn 3 dual-line
number 6002
ephone-dn 4 octo-line
number 6003
ephone 2
Mac-address Y.Y.Y.y
button 1:3 2:4
Single Line
Example:
ephone-dn 1 octo-line
number 4001 no-reg both
description +85224024001
name SiteC Phone 1
huntstop channel 1
7. 777
EPHONE – DN Single Line
• Single-Line
–Makes one call connection at a time using one phone
line button. A single-line directory number has one
telephone number associated with it.
–Should be used when phone buttons have a one-to-one
correspondence to the PSTN lines that come into a
Cisco Unified CME system.
–Should be used for lines that are dedicated to intercom,
paging, message-waiting indicator (MWI), loopback,
and music-on-hold (MOH) feed sources.
–Must have more than one single-line directory number
on a phone when used with multiple-line features like
call waiting, call transfer, and conferencing.
–Can be combined with dual-line directory numbers on
the same phone.
ephone-dn 2
number 6001
ephone 1
Mac-
button 1:2
8. 888
EPHONE – DN Dual
• Has one voice port with two channels.
• Supported on IP phones that are running SCCP; not
supported on IP phones that are running SIP.
• Can make two call connections at the same time using one
phone line button. A dual-line directory number has two
channels for separate call connections.
• Can have one number or two numbers (primary and
secondary) associated with it.
• Should be used for a directory number that needs to use
one line button for features like call waiting, call transfer,
or conferencing.
• Cannot be used for lines that are dedicated to intercom,
paging, message-waiting indicator (MWI), loopback, and
music-on-hold (MOH) feed sources.
• Can be combined with single-line directory numbers on
the same phone.
ephone-dn 3 dual-line
number 6001
ephone 1
Mac-
button 1:3
9. 999
EPHONE – DN Octo Line
• An octo-line directory number supports up to eight active
calls both incoming and outgoing, on a single button of a
SCCP phone
• An octo-line directory number can split its channels
among other phones that share the directory number
• Octo-line directory number can handle multiple calls
because octo-line directory numbers do not require a
different ephone-dn for each active call
• After a phone answers an incoming call, the answering
phone is in the connected state. Other phones that share
the octo-line directory number are in the remote-in-use
state
• After a connected call on an octo-line directory number is
put on-hold, any phone that shares this directory number
can pick up the held call
• The Barge and Privacy features control whether other
phones are allowed to view call information or join calls on
the shared octo-line directory number.
ephone-dn 3 Octo-line
number 6001
ephone 1
Mac- X.X.X
button 1:3
10. 101010
Phones in Cisco Unified CME
ephone
Ethernet Phone
Is a software configuration for a phone in Cisco Unified CME
Cisco Unified CME 8.8 and later versions support the following phones:
• Cisco Unified 3905 SIP IP Phones
• Cisco Unified 6901 and 6911 SIP IP Phones
• Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones
• Cisco Unified 8941 and 8945 SIP IP Phones
• Cisco Unified 6945, 8941, and 8945 SCCP IP Phones
ephone 1
mac-address 0023.339D.F9EB
username "scone" password cisco
type 7965
button 1:2
Example: Extension 4001
ephone-dn 2 octo-line
number 2002 no-reg both
description +85224022002
name SiteC Phone 1
huntstop channel 1
12. 121212
CUCEM – SIP Based
• Enable SIP Service
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0.102
bind media source-interface GigabitEthernet0/0.102
registrar server expires max 1200 min 300
voice register global
mode cme
source-address 142.102.66.254 port 5060
max-dn 5
max-pool 5
authenticate register
authenticate realm all
timezone 42
date-format D/M/Y
voicemail 4220
ntp-server 157.26.1.250 mode unicast
conference hardware
camera
video
voice register dn 1
number 4002
name SC Phone 2
label +85224024002
mwi
voice register pool 1
id mac B414.89A2.F83A
type 9951
number 1 dn 1
description SC Phone 2
no vad
camera
video
15. 151515
Voice Network Directory
• Local Directory Service
• Name command under ephone-dn can be searchable from this local directory
• Names are used both for building the internal corporate phone directory (often called the local
directory) and for caller ID information
• By Default CME - organizes the local directory alphabetically by first name
• Also add manual entries to the directory by using the directory entry command
• You can add up to 100 manual entries to the local CME directory
CME_Voice(config-telephony)# directory ?
entry Define new directory entry
first-name-first first name is first in ephone-dn name field
last-name-first last name is first in ephone-dn name field
CME_Voice(config-telephony)# directory last-name-first
CME_Voice(config-telephony)# directory entry ?
<1-100> Directory entry tag
clear clear all directory entries
CME_Voice(config-telephony)# directory entry 1 ?
WORD A sequence of digits representing dir. number
CME_Voice(config-telephony)# directory entry 1 1599 ?
name Define directory name
CME_Voice(config-telephony)# directory entry 1 1599 name ?
LINE A string - representing directory name (max length: 24 chars)
CME_Voice(config-telephony)# directory entry 1 1599 name Corporate Support
16. 161616
Configuring Call Forwarding
• Forward calls to a different destination
• To forward calls from the IP phone, just press the CFwdAll softkey button
• Forwarding calls from the command line gives you more options than does
forwarding callsfrom the IP phone
• Tandem hop in the call flow is the default call flow. When call is forwarded, Voice
traffic goes through the forwarding station thus quality can suffer
• H.450.3 standard represents a method that allows the CME router to redirect the call
directly to the final destination instead of acting as a tandem hop
ephone-dn 1
call-forward busy 4001
call-forward noan 4001 timeout 18
17. 171717
Configuring Call Transfer
• Transferring calls is another basic requirement of a business
phone system
• To transfer a call, press the Trnsfer softkey while on an active
call
• Two Type of Transfer
–Consult: Consult transfer allows you to speak with the other party
before transferring the call
–Blind:Blind transfer immediately transfers the call after you dial the
number
telephony-service
transfer-system full-consult
transfer-pattern 5...
transfer-pattern 4...
18. 181818
Configuring Call Park
• Allowing you to retrieve the call from any phone in the organization once it is parked
by pressing parks softkey
• Any IP phone that can dial the park extension number can retrieve the call
• Call park system works by finding free ephone-dns where call will be held
temporarily. From the retrievable phone, just dial th ephone-dn number
• Calls being parked at random extensions might work well for a warehouse
environment with a voice-paging system
ephone-dn 15
number 7001
park-slot timeout 30 limit 10
ephone 5
button 1:11 2:12 3:13
transfer-park blocked
19. 191919
Feature Call Pickup
• Call Pickup allows a phone user to answer a call that is ringing on another phone
–Directed Call Pickup—Call pickup, explicit ringing extension. Any local phone user can
pick up a ringing call on another phone by pressing a soft key and then dialing the
extension.
–Group Pickup, Different Group—Call pickup, explicit group ringing extension. A phone
user can answer a ringing phone in any pickup group by pressing the GPickUp soft key and
then dialing the pickup group numbe
–Local Group Pickup—Call pickup, local group ringing extension. A phone user can pick up
a ringing call on another phone by pressing a soft key and then the asterisk (*) if both
phones are in the same pickup group.
ephone-dn 1
number 4001
pickup-group 100
ephone-dn 2
number 4002
pickup-group 100
ephone-dn 3
number 4003
pickup-group 300
20. 202020
Feature Template
• Phone Templates
–An ephone template is a set of features that can be applied to one or more individual
phones using a single command.
–Templates allow you to uniformly and easily implement the features you select for a set of
phones.
• Ephone-dn Templates
–Ephone-dn templates allow you to apply a standard set of features to ephone-dns
ephone-template 15
features blocked Park Trnsfer
ephone-dn 2
number 2333
ephone 36
button 1:2
ephone-template 15
ephone-dn-template 3
call-forwarding busy 4000
call-forwarding noan 4000 timeout 30
pickup group 4
ephone-dn 23
number 2323
ephone-dn-template 3
ephone-dn 33
number 3333
ephone-dn-template 3
21. 212121
Feature - Paging
• A paging number can be defined to relay audio pages to a group of designated phones
• When a caller dials the paging number (ephone-dn), each idle IP phone that has been
configured with the paging number automatically answers using its speakerphone mode
• Displays on the phones that answer the page show the caller ID that has been set using
the name command under the paging ephone-dn
• Audio paging provides a one-way voice path to the phones that have been designated to receive
paging
Ephone-dn 10
number 8888
paging ip 239.1.1.10 port 2000
Ephone 1
paging-dn 10 multicast
Ephone 2
paging-dn 10 multicast
22. 222222
Single Number Reach in CME
• The Single Number Reach (SNR) feature allows users to answer incoming calls to their
extension on either their desktop IP phone or at a remote destination, such as a mobile phone
• Users can pick up active calls on the desktop phone or the remote phone without losing the
connection
• This enables callers to dial a single number to reach the phone user. Calls that are not answered
can be forwarded to voice mail
• Remote destinations may include the following devices:
–Mobile (cellular) phones
–Smart phones
–IP phones not belonging to the same Cisco Unified CME router as the desktop phone
–Home phone numbers in the PSTN. Supported PSTN interfaces include PRI, BRI, SIP,
and FXO
• Cisco Unified CME rings the desktop IP phone first. If the IP phone does not answer within
the configured amount of time
Mobile (cellular) phones.
•
23. 232323
Single Number Reach in CME
ephone-template 1
softkeys idle Dnd Gpickup Pickup Mobilit
softkeys connected Endcall Hold LiveRcd Mobility
ephone-dn 10
number 6001
mobility
Snr 94163013001 3 delay 5 timeout 15 cfwd-noan 4400
Softkey template allows you to define a softkey
button which can be use to enable or disable Mobility
24. 242424
CCME ephone-hunt
ephone-hunt allows CCME administrators to:
• Define a pilot number for a hunt group
• Sequential mode: specifies an ordered list of extension
numbers to sequentially hunt through
• Peer mode: specifies a random start point in a circular
list of extension numbers
• Longest Idle: specifies who is idle for long.
• Define a final destination to forward the call to if the call
is not answered or all members are busy
25. 252525
CCME Hunting
Ephone-hunt 1 seq
pilot 6500
list 6001, 6002
final 6000
timeout 5
ephone-hunt 2 peer
pilot 6000
list 6002, 6001, 6003
final 3001 can not be 6500
preference 1
timeout 30
no-reg
09:00 06/500/05
6001
6001
VoiceBootcamp Inc.
09:00 06/500/05
6001
6001
VoiceBootcamp Inc.
IP phone 1
IP phone 2
Inbound call to 6500
If 6001 is busy
and/or not
answering
26. 262626
End of Chapter
• CME is designed for small business
• Can be manage by GUI or Command Line
• Integrated GUI or CCP